The invention relates to a method of packet-oriented transmission of data between at least two communication devices, and also a control device and a conversion device.
Communication connections for speech have heretofore predominantly been constructed as connection-oriented. A physical connection is exclusively provided for this purpose, for a signal transmission between two communication end points, and is reserved for the whole time of the connection. This is also termed line-oriented transmission, static routing, or circuit switching.
With the advent of packet-oriented data networks (packet switching), such as for example the Internet, communication is offered in the fixed-network range which provides cost reduction in comparison with connection-oriented telecommunication. This is in particular to be ascribed to the efficient capacity utilization of a connection, since in contrast to circuit-switching, packet-switching does not occupy the physical transmission medium for the whole duration of the connection. The data to be transported is divided into individual data packets, each data packet receiving an address code which identifies the receiver of the transmission. The individual data packets are then transmitted independently of each other—and they can even use different transmission paths. The principle of packet switching is defined in various standards; the best-known standard is described in ITU-T Recommendation x.25.
VoFR (Voice over Frame Relay) or VoIp (Voice over IP) are for example known as packet-oriented transmission methods. Here the speech data are digitized, undergo a source coding and preferably a channel coding, and are divided into data packets, which are then transmitted over the Internet. VoIp in particular is predicted to be of considerable importance for future speech communication. Here the speech is digitized and compressed (source coded) using software or hardware, the compressed speech then representing the use data region of the IP packets. Selected call numbers are converted into IP addresses and placed in the IP header as target information. The IP packets are now transported over plural network nodes distributed in the data network, as far as the distant end of the speech connection. The distant end memorizes the incoming packets and combines then again in the correct sequence. If a packet is damaged or lost, it is not sent anew. The speech information is taken out of the packets at the distant end and is then supplied to a coding device in which the information is then inversely source- and channel-coded and is then made audible by suitable hardware.
Since a delayed arrival of adjacent IP speech packets (delay) has a negative effect on the quality of the speech connection, various methods of time synchronizing, and thus minimizing the delay, are provided in the IP network in order to maintain the QoS (Quality of Service). The insertion of a Real Time Protocol (RTP) belongs to this method, in which each IP packet, etc., additionally receives a time stamp with the time of origin and a sequence number (sequence information). This permits the receiving device to combine packets not only in the correct sequence, but also time-synchronized. The RTP furthermore defines the coding of audio signals according to G.723, G.711 or G.729. Concerned here are encoding and decoding methods (codecs), which are defined by the ITU for analog and digital encryption of speech in telephone networks.
G.711 about corresponds to the ISDN standard, in which speech data are transmitted with a data stream of 64 kBit/s. By additional source coding, the data rate can be reduced as far as 9.6 kBit/s, making transmission over VoIp networks possible for the first time.
The widespread CELP method (Codebook Excited Linear Predictive Coding) method counts as one of these methods; it processes human speech with a complicated mathematical model. The output of this source coding is a data stream with a data rate of 16 kBit/s, and the speech quality almost corresponds to ISDN speech quality. Combined with a Dual Rate Speech Coding defined in the G.723 standard, even a data stream with a data rate of only 5.3 kBit/s is sufficient, with a reduced but subjectively accepted speech quality. Besides a lower network loading, this brings the further advantage that plural IP packets are buffered without endangering the real time condition. The quality of the speech transmission on the Internet thus rises with a falling data rate for a speech channel.
Voice over IP likewise includes protocols for the transmission of different multimedia formats, which go beyond pure speech data. In particular, the possibility of transmitting video conferences falls within this expansion.
The MPEG-1 standard counts as the best known codec for source coding video signals. Here the resolution of the coded picture is limited to a Source Input Format (SIF), in which the chrominance is under-sampled in both directions, and the number of pixels is halved. The video data is reduced by motion estimation and redundancy reduction such that transmission over the Internet is made possible.
The H.261 standard, adopted by the CCITT in 1990, represents a widely diffused standard for source coding of video material. This compression standard was developed for videoconferencing and other video services in ISDN at bit rates of multiples of 64 kBit/s. The improved and extended standard H.263 was developed later, and was specified for bit rates smaller than 64 kBit/s.
Thus there are many characteristic codecs for each data category (audio or video). For communication, of course, all potential communication partners must have a command of the respective codec with which information was coded, in order to be able to perform subsequent decoding. For example, if a speech message is encoded with the audio codec G.729 and transmitted to a receiving device which cannot decode this codec, the call setup will be broken off.
With increasing propagation of different VoIp networks and application scenarios, special equipments more and more frequently come to be used, which only have a command of special coding methods. In such network configurations, an increasing number of coding methods has to be implemented for numerous standard end equipments, entailing higher computing power and higher license costs.
One potential object of the invention therefore is to develop a process of this category to the effect that a call can be set up, even with differently furnished coding and decoding methods. A further potential object of the invention is to make available a control device and a converting device for performing the method.